EP1853005A1 - Verfahren und Vorrichtung zur Ubetragung von Information über einem Netz zu Bestimmungsortvorrichtungen - Google Patents

Verfahren und Vorrichtung zur Ubetragung von Information über einem Netz zu Bestimmungsortvorrichtungen Download PDF

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Publication number
EP1853005A1
EP1853005A1 EP06405189A EP06405189A EP1853005A1 EP 1853005 A1 EP1853005 A1 EP 1853005A1 EP 06405189 A EP06405189 A EP 06405189A EP 06405189 A EP06405189 A EP 06405189A EP 1853005 A1 EP1853005 A1 EP 1853005A1
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EP
European Patent Office
Prior art keywords
network
digital signal
destination
unit
source unit
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Legal status (The legal status is an assumption and is not a legal conclusion. Google has not performed a legal analysis and makes no representation as to the accuracy of the status listed.)
Withdrawn
Application number
EP06405189A
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English (en)
French (fr)
Inventor
Thierry Heeb
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Anagram Technologies SA
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Anagram Technologies SA
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Publication date
Application filed by Anagram Technologies SA filed Critical Anagram Technologies SA
Priority to EP06405189A priority Critical patent/EP1853005A1/de
Priority to US11/739,694 priority patent/US20070252730A1/en
Publication of EP1853005A1 publication Critical patent/EP1853005A1/de
Withdrawn legal-status Critical Current

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    • H—ELECTRICITY
    • H03—ELECTRONIC CIRCUITRY
    • H03F—AMPLIFIERS
    • H03F3/00—Amplifiers with only discharge tubes or only semiconductor devices as amplifying elements
    • H03F3/20—Power amplifiers, e.g. Class B amplifiers, Class C amplifiers
    • H03F3/21—Power amplifiers, e.g. Class B amplifiers, Class C amplifiers with semiconductor devices only
    • H03F3/217—Class D power amplifiers; Switching amplifiers
    • H03F3/2173—Class D power amplifiers; Switching amplifiers of the bridge type
    • G—PHYSICS
    • G10—MUSICAL INSTRUMENTS; ACOUSTICS
    • G10L—SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
    • G10L19/00—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis

Definitions

  • the present invention relates to a method and a network-based system that allow transferring information content, particularly audio information, or, more generally, information relating to recorded analogue signals, from a centralised source unit, particularly an audio source unit, in digital form over at least one network to at least one destination unit, particularly to an audio destination unit, in which the transferred data are further processed to provide an analogue signal.
  • the present invention relates in particular to a network-based system that allows transferring audio information, which is available in the audio source unit in any digital or analogue format, in a digital format over the network to at least one audio destination unit, where the audio information is forwarded to a switching stage of a digital audio amplifier, typically a class-D amplifier or a low pass filter followed by an analogue amplifier.
  • a digital audio amplifier typically a class-D amplifier or a low pass filter followed by an analogue amplifier.
  • the present invention further relates to a network-based system that allows transferring digital audio information over a wired network such as a wired bus system or a switched network by means of a connection-oriented service, e.g. in a circuit switched network, or by means of a connectionless service, e.g. in a packet switching network such as an IP-network or a wireless optical or radio frequency network.
  • a connection-oriented service e.g. in a circuit switched network
  • a connectionless service e.g. in a packet switching network such as an IP-network or a wireless optical or radio frequency network.
  • the digital signal can be filtered, modulated, and thus brought into various forms suitable for transmission over a specific network.
  • the first stage performs interpolation of a Nyquist rate PCM (pulse coded modulation) signal to produce an oversampled PCM digital signal.
  • PCM pulse coded modulation
  • This signal is then fed into a modulation circuit.
  • the modulation is typically sigma delta ( ⁇ - ⁇ ) or pulse width modulation (PWM) and thus produces a final signal that has only a finite number of voltage levels, usually two or three.
  • the key aim of the modulation is to produce a signal where the information of the PCM signal value is represented by the mean-value of the bit stream. This allows the original signal information to be extracted by a simple analogue low pass filter, which can be constructed from passive components like inductors and capacitors. [2], figure 2 shows how the mean value can be represented by digital pulses.
  • a wide range of low bandwidth, high resolution, ⁇ - ⁇ ADCs (series AD 77XX) is available for example from Analog Devices Inc. (http://www.analog.com/en/index.html).
  • these devices typically contain current sources, a multiplexer, a PGA and analogue input buffers on board.
  • D/A and A/D Converters are further described in [3], Richard C. Dorf, The Electrical Engineering Handbook, 2nd Edition, CRC Press, 1997 Boca Raton, chapter 32, page 841 ff .
  • the two or three level modulated signals can have a variable width and a variable or fixed cycle-time or the width can be fixed and there is no specific cycle-time.
  • a large width or a high number of pulses represents a greater input value than a small pulse-width or a low number of pulses.
  • Amplifiers for these two or three level signals belong to the so-called class-D amplifiers, because the power stages have no linear area of operation.
  • the PCM-signal can be forwarded from a signal source 1, e.g. a signal processor 10, over a bus system or a wired network to a local amplifier 3L and over a circuit switched network or packet switching network 2 to a distant digital amplifier, which comprises said Pulse-Width or Sigma-Delta Modulator (the drawings show both alternatives and examples for the PWM- and ⁇ - ⁇ - modulated signals), from which the modulated signals are forwarded to a switching stage 32, and used there to activate at least one pair of switches that are connected on the one side to supply voltages V+, V- and on the other side to a low pass filter 33 that is connected to a load such as a loudspeaker 34.
  • a signal source e.g. a signal processor 10
  • a signal source e.g. a signal processor 10
  • a signal source e.g. a signal processor 10
  • a bus system or a wired network to a local amplifier 3L and over a circuit switched network or
  • the amplifiers 3L and 3d shown in figure 1 are class-D amplifiers, which can be designed to provide the same performance as known class A / B / A-B amplifiers.
  • the volume of the digital power amplifier can be controlled by controlling the level of the supply voltages V+, V-.
  • Digital audio amplifiers were introduced on the market about thirty years ago (e.g. Sony TA-N88). Modern digital audio amplifiers such as the Sharp SM-SX 100 achieve very high performance levels.
  • FIG. 2 shows a network-based system designed for transferring audio information.
  • This system allows distributing original audio information in PCM signals or encoded or compressed PCM signals (i.e. by using MP3, WMA, AAC, etc compression encoding and decoding) issued by a first network terminal, e.g. a content & DRM server 10 provided in an audio source unit 1, over a packet switching network 2, e.g. an IP-network, to a number of network terminals 21, e.g. personal computers, which forward the received audio information, preferably provided in PCM format to digital power amplifiers 3A, 3B or to D/A converters followed by an analogue amplifier.
  • the modulators provided in the digital power amplifiers 3A, 3B transform the received PCM signals into corresponding streams of binary data.
  • Content information is further forwarded over a mobile telecommunication network PLMN, e.g. operating according to GSM or UMTS standards, to mobile terminals.
  • PLMN mobile telecommunication network
  • the network terminal 21B further forwards the received original audio information in PCM or encoded format to a handheld audio terminal 350, which typically is equipped with a large storage media, such as hard or fixed disk and which is designed to deliver an analogue audio signal to ear phones or an analogue audio signal or a digital PCM-signal further to the analogue or digital input of the audio amplifier 3A.
  • a handheld audio terminal 350 typically is equipped with a large storage media, such as hard or fixed disk and which is designed to deliver an analogue audio signal to ear phones or an analogue audio signal or a digital PCM-signal further to the analogue or digital input of the audio amplifier 3A.
  • the original audio information can illegally be accessed or distributed or copied to other storage media without the control or consent of the owner of the audio information, thus often violating regulations of copyright protection law.
  • owners and lawful distributors it would be desirable for owners and lawful distributors to find a solution to this problem on a technical and not on a legal basis.
  • DRM Digital Rights Management
  • scrambled digital content is descrambled within a subscriber terminal device, by conducting digital rights management by a first component of the subscriber terminal device to determine whether entitlements needed for accessing digital content in a descrambled format are available to the subscriber terminal device; and deriving a key within a second component of the subscriber terminal device logically separate from and independent of the first component in response to determining that all of the entitlements are needed for accessing the digital content are available to the subscriber terminal device, the key being used to decrypt service keys used for descrambling the scrambled digital content.
  • the DRM-measures cover the network-based system shown in figure 2 from the content server to the subscriber terminals 21, 350.
  • the path from the subscriber terminals 21, 350 to the audio amplifier remains unprotected.
  • the inventive method and the inventive network-based system allow transferring information, particularly audio information from a centralised source unit, particularly an audio source unit, in digital form over a network to at least one destination unit, particularly to an audio destination unit, in which the transferred data are further processed to provide an analogue signal.
  • the present invention allows transferring information derived from an analogue signal over a network, e.g.
  • the second digital signal can be applied in the destination unit to the switching stage of a digital power amplifier that is connected to a low pass filter, which provides an output signal to a loudspeaker.
  • analogue signals may also be used for other purposes, e.g. for controlling networked systems particularly distributed systems.
  • Concentrating the modulators, which are normally part of a digital amplifier, in the centralised source unit provides several advantages.
  • the decentralised destination units do not require signal processing units.
  • all mathematical modifications of the digital signals are performed in the centralised source unit, where corresponding processing hardware and software is readily available.
  • centrally expanding the processing power results in small cost increases compared to decentrally establishing processing capabilities in the destination units.
  • the digital signals can therefore be processed in the centralised source unit with high accuracy at low cost. As a result overall cost of the network-based system can significantly be reduced.
  • the second digital signal that is produced by an individual modulator by means of pulse width modulation or sigma-delta modulation or a derivative thereof or by means of a sample rate conversion is not available in a suitable form for copying, although it may be presented in the highest possible quality.
  • unlawful users will not have access to the original content, e.g. audio content, along the whole transmission chain. Access to transferred content would practically be available in the analogue domain at the output of the low pass filter only.
  • Bit streams of the second digital signals produced by the modulators contained in the centralised source unit can be multiplexed, encoded, encrypted and/or packetised in the centralised source unit and de-multiplexed, decoded, decrypted and/or de-packetised in the destination unit according to any known method, as described for example in [6].
  • control frames can be introduced into the packet stream with little effort, thus allowing centralisation of the control circuitry in the centralised source unit, thus further cutting down expenses for decentralised circuitry.
  • the bit stream of the second digital signal is forwarded to a buffer such as a FIFO-buffer. If the buffer is large enough and the drifts of the clock frequencies are small and the clock frequencies are identical in average no overflow will occur. However, if the frequencies constantly deviate a correction is required which can preferably be performed as follows.
  • the clock frequency of the centralised source unit can be recovered in the destination unit, preferably by means of a phase locked loop. Hence, with this method, the clock frequency of the destination unit is firmly locked to or following the clock frequency of the centralised source.
  • the content level in the buffer is measured and a corresponding control signal is sent to a sample rate converter provided in the centralised source unit, which adjusts the clock frequency for the related modulator in such a way that the content level in the buffer remains within a given range.
  • a common network time and/or clock may advantageously be used for synchronising all relevant system clocks in the centralised source unit and the individual destination units.
  • delays occurring in the network-based system can be measured and compensation in order to maintain a phase coherent operation for all system parts.
  • Phase coherent operation may also be guaranteed by using a common system time that is locally maintained in clock units or timers and that allows synchronisation of the play out of the individual bit streams or allowing calculating different start times in order to compensate for propagation delays outside the network-based system (e.g. in the event that a loudspeaker has been placed at a place that is farther a part from the audience than the other loudspeakers, which will be assigned a corresponding delay.
  • the start time can be calculated in the centralised source unit in advance.
  • a suitable common clock time can be determined in the centralised source unit in advance based on known delays that have been measured within the network-based system.
  • the determined common clock time can be forwarded to the destination units within a frame such as a header contained in the bit stream of the second digital signal or over a separate control channel.
  • a suitable common clock time can be determined after the start of the transfer of the bit stream of the second digital signal, and after the detection of a header in the destination units, that has been signalled to the centralised source unit, which then forwards the determined common clock time to the destination units over a separate control channel.
  • the network-based system can be used on the one hand particularly for private or public audio or multimedia entertainment systems and on the other hand for commercial or non-commercial distribution systems, particularly DRM-distribution systems.
  • the application of the inventive solution is scalable from a small home application to a global application with unlimited subscribers.
  • Figure 3 shows an inventive network-based system with a centralised source unit 1 comprising at least one modulator 131, which modulates a first digital signal d11 ; d12 that has been provided from an audio processor 10 and which forwards a corresponding second (modulated) digital d21; d22 to a first and to a second destination unit 3L, 3D, in which the digital signal is forwarded to a digital audio amplifier consisting of a switching stage 32, and a low pass filter 33 or, as an alternative, to a preferably active low pass filter that provides a low level analogue signal to an analogue amplifier.
  • a centralised source unit 1 comprising at least one modulator 131, which modulates a first digital signal d11 ; d12 that has been provided from an audio processor 10 and which forwards a corresponding second (modulated) digital d21; d22 to a first and to a second destination unit 3L, 3D, in which the digital signal is forwarded to a digital audio amplifier consisting of a switching
  • a modulator 131 with individual modulator modules is provided in the centralised source unit 1, which may be implemented within a single computer system 1. Since the computing power of modern computer systems 1 is extremely high, said modulators 131 or 1311, 1312 can be realised with high quality at low cost, thus avoiding the replication of corresponding processing units in the peripheral destination units.
  • the modulation performed by the modulators 131 or modulator modules 1311, 1312 is preferably a pulse width modulation or a sigma-delta modulation or a derivative thereof. Both alternatives are illustrated in figure 3.
  • the resulting second digital signal which directly represents the audio content can be multiplexed, encoded, encrypted and/or, if required, packetised according to a known method and then transferred over the network to the destination units 3. Packetising of encoded and encrypted data is described for example in [7], US 6'990'202 .
  • FIG. 5 shows the inventive network-based system of figure 4, with a clock recovery unit, which comprises a phase locked loop PLL integrated in the clock unit 37 of the destination unit 3D.
  • Phase Locked Loops PLL which comprise a phase detector PD, filter LF and a voltage controlled oscillator VCO are described in [3], chapter 76, pages 1727-1734.
  • the clock frequency at the input of the modulator is fs1.
  • the clock frequency fs2 n * fs1 appears.
  • the average rate at which data is entering the buffer 36 is kept identical to the rate at which data is leaving the buffer 36, this avoiding an overflow.
  • Figure 6 shows the inventive network-based system of figure 4, with a clock and measuring unit 37 provided in the destination unit 3D, which provides a fixed clock signal fs3 to the buffer 36 and which measures and signals the content level of the buffer 36 to the clock unit 17 of the centralised source unit 1 or directly to a sample rate converter 15 therein, which adjusts the sampling frequencies of the related modulator 131 in order to keep the content level within a predetermined range.
  • a clock and measuring unit 37 provided in the destination unit 3D, which provides a fixed clock signal fs3 to the buffer 36 and which measures and signals the content level of the buffer 36 to the clock unit 17 of the centralised source unit 1 or directly to a sample rate converter 15 therein, which adjusts the sampling frequencies of the related modulator 131 in order to keep the content level within a predetermined range.
  • sampling rate As described in [1], chapter 2.4, page 46, generally, there is only one sampling rate used within a digital signal-processing system. Sampling is the process of converting a signal into a numeric sequence (a function of discrete time or space).
  • the Nyquist-Shannon theorem states conditions under which the samples represent no loss of information and can therefore be used to reconstruct the original signal with arbitrarily good fidelity. It states that the signal must be band limited and that the sampling frequency must be at least twice the signal bandwidth. According to [1] a change of the sampling frequency may be desirable under certain conditions. If the bandwidth of the signal is reduced a reduction of the sampling rate saves computing time. In the contrary, the sampling rate is increased, if the bandwidth is expanded or if interpolation efforts shall be reduced.
  • Figure 7 shows the inventive network-based system of figure 4, with a network time base control system 2000 that synchronises local clocks and times in the centralised source unit 1 and in the destination units 3.
  • Maintaining Synchronization in a Network is described for example in [8], WO 00/0048367 .
  • network nodes participate in a scheme to maintain synchronization in the network. Such maintenance is needed to compensate for different "drift" rates for each node's timer(s).
  • each node's local Network Time is only adjusted in the forward direction, and the network will roughly track the time of the node in the network that happens to have the fastest clock. The Network Time will switch to tracking the time of the node with the next fastest clock.
  • Timers 371 provided in the in the individual destination units 3 and preferably also in the centralised source unit 1 are set to and kept at the same time indication.
  • the start of reading data from the buffers 36 in the destination units 3 can be set to the same time, thus avoiding phase shifts between the signals emitted from the destination units.
  • it may normally be optimal to set the start of the reading process for all destination units 3 at the same time setting it may be advantageous to use different time settings, if the propagation delays outside the network-based system, e.g. within a large concert hall are not balanced.
  • the network-based system shown in figure 7 comprises a delay measuring unit 6, which allows measuring all relevant system delays.
  • the delay measuring unit 6 is preferably located in the centralised source unit 1 and provides the obtained delay information to a process which calculates time settings.
  • a suitable common clock time can be determined in the centralised source unit 1 in advance based on the reported delays.
  • the determined common clock time can be forwarded to the destination units in a frame such as a header contained in the bit stream of the second digital signal or over a separate control channel.
  • a suitable common clock time can be determined after the start of the transfer of the bit stream of the second digital signal.
  • a header is provided in front of the modulated bit stream, which header can be detected in the destination units and signalled to the centralised source unit 1, which then forwards the determined common start time to the destination units over a separate control channel.
  • Figure 7 shows a header in front of the PWM-modulated bit stream, which may contain a start time or which may simply be used for the detection and signalling of the receipt of the bit stream.
  • the PWM-modulated bit stream is forwarded to a detector 370, which detects the receipt of the header and, if present, unpacks the predetermined time which is forwarded to the timer 371.
  • sampling rate converter 15 which reports the sampling rate selected to the destination units 3, if this information, e.g. a fixed factor, is not already available there.
  • the network time base & control system 2000 can be regarded as a separate layer of the inventive network-based system, with timing control signals and timing messages exchanged between nodes that represent the central and peripheral units 1, 3 of the system.
  • the local times TIME1, ..., TIME4 and delays DELAY1, ..., DELAY4 may be registered and continuously updated, corrected and synchronised.
  • Figure 8 shows symbolically the multiplexing architecture of an inventive network-based system comprising a centralised source unit 1 with a multiplexer 18 and four destination units 3D1, ..., 3D4 with de-multiplexers 301, ..., 304.
  • the multiplexer 18 multiplexes four different bit streams received from modulators 131A, ..., 131D and forwards the resulting bit stream to the demultiplexers 301, ..., 304, which extract the part from the bit stream which is dedicated to the related destination unit 3.
  • the multiplexed bit stream can be encoded, packetised or compressed.
  • Figure 9 shows the inventive network-based system of figure 7 with a central control unit 1000 provided in the centralised source unit 1 that communicates with the local control unit 3000 of the destination units 3D via a control channel ctrl that is separate from the content channel.
  • the central control unit 1000 preferably allows controlling of all relevant functions and parameters used in the destination unit.
  • the central control unit 1000 may for example control the power supply 391, particularly the supply voltages provided to the switching unit 32 and an input and/or output device 392 such as a display in the destination unit 3.
  • Figure 10 shows the inventive network-based system of figure 7 with the central control unit 1000 provided in the centralised source unit 1 forwarding control signals in-band or interleaved within the packetised modulated bit stream to the local control unit of the destination units 3D.
  • an interleaver 100 is provided in the centralised source unit 1, which encloses control signals, e.g. in the packetised stream of modulated content.
  • a de-interleaver is provided, which separates the control data stream and the content stream, which is forwarded to the buffer 36, and if required to the clock unit 37 for detecting start signals or start timings or for any recovery purposes.
  • Figure 11 shows a sample rate converter 15/131 provided in the centralised source unit 1 that is used for sample rate conversion and or signal modulation.
  • the sampling rate converter 15/131 is typically operated in high frequency ranges in order to keep quantisation errors and interpolation efforts as low as possible.
  • Converted digital signals are forwarded to the buffer 36, which provides the above described services, in destination unit 3 and further to an digital-to-analog converter 310.
  • Figure 12 shows the network-based system of figure 2 after the implementation of the inventive solution. It is illustrated that the content is protected by means of the inventive solution without any gap, on the whole transmission path from the centralised source unit 1 to the load, provided in the destination units.

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  • Engineering & Computer Science (AREA)
  • Power Engineering (AREA)
  • Computational Linguistics (AREA)
  • Signal Processing (AREA)
  • Health & Medical Sciences (AREA)
  • Audiology, Speech & Language Pathology (AREA)
  • Human Computer Interaction (AREA)
  • Physics & Mathematics (AREA)
  • Acoustics & Sound (AREA)
  • Multimedia (AREA)
  • Compression, Expansion, Code Conversion, And Decoders (AREA)
  • Data Exchanges In Wide-Area Networks (AREA)
EP06405189A 2006-05-01 2006-05-01 Verfahren und Vorrichtung zur Ubetragung von Information über einem Netz zu Bestimmungsortvorrichtungen Withdrawn EP1853005A1 (de)

Priority Applications (2)

Application Number Priority Date Filing Date Title
EP06405189A EP1853005A1 (de) 2006-05-01 2006-05-01 Verfahren und Vorrichtung zur Ubetragung von Information über einem Netz zu Bestimmungsortvorrichtungen
US11/739,694 US20070252730A1 (en) 2006-05-01 2007-04-25 Method and network-based system for transferring information over a network to destination devices

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Application Number Priority Date Filing Date Title
EP06405189A EP1853005A1 (de) 2006-05-01 2006-05-01 Verfahren und Vorrichtung zur Ubetragung von Information über einem Netz zu Bestimmungsortvorrichtungen

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Cited By (1)

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CN103905834A (zh) * 2014-03-13 2014-07-02 深圳创维－Rgb电子有限公司 音频数据编码格式转换的方法及装置

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CN106937001A (zh) * 2017-04-20 2017-07-07 维沃移动通信有限公司 一种音频保存方法和移动终端

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US5898340A (en) 1996-11-20 1999-04-27 Chatterjee; Manjirnath A. High power efficiency audio amplifier with digital audio and volume inputs
US20020072816A1 (en) 2000-12-07 2002-06-13 Yoav Shdema Audio system
EP1411517A1 (de) * 2002-10-18 2004-04-21 Motorola, Inc. Kontrolleinheit und Verfahren zur Übertragung von Tonsignalen über ein optisches Netzwerk
US20040157548A1 (en) 2003-02-06 2004-08-12 Eyer Mark Kenneth Home network interface legacy device adapter
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US5898340A (en) 1996-11-20 1999-04-27 Chatterjee; Manjirnath A. High power efficiency audio amplifier with digital audio and volume inputs
US20020072816A1 (en) 2000-12-07 2002-06-13 Yoav Shdema Audio system
EP1411517A1 (de) * 2002-10-18 2004-04-21 Motorola, Inc. Kontrolleinheit und Verfahren zur Übertragung von Tonsignalen über ein optisches Netzwerk
US20040157548A1 (en) 2003-02-06 2004-08-12 Eyer Mark Kenneth Home network interface legacy device adapter
US20060044057A1 (en) 2004-08-26 2006-03-02 Rahmi Hezar Class-D amplifier having high order loop filtering

Cited By (1)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
CN103905834A (zh) * 2014-03-13 2014-07-02 深圳创维－Rgb电子有限公司 音频数据编码格式转换的方法及装置

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